Traditionally, telephony communications within the United States were handled by the public switched telecommunications network (PSTN). The PSTN can be characterized as a network designed for voice communications, primarily on a circuit-switched basis, with full interconnection among individual networks. The PSTN network is largely analog at the local loop level, digital at the backbone level, and generally provisioned on a wireline, rather than a wireless, basis. The PSTN includes switches that route communications between end users. Circuit switches are the devices that establish connectivity between circuits through an internal switching matrix. Circuit switches set connections between circuits through the establishment of a talk path or transmission path. The connection and the associated bandwidth are provided temporarily, continuously, and exclusively for the duration of the session, or call. While developed to support voice communications, circuit switches can support any form of information transfer (e.g., data and video communications).
In an effort to increase the amount and speed of information transmitted across networks, the telecommunications industry is shifting toward broadband packet networks designed to carry a variety of services such as voice, data, and video. For example, asynchronous transfer mode (ATM) networks have been developed to provide broadband transport and switching capability between local area networks (LANs) and wide area networks (WANs). One technology used to provide high-speed data transfer services to individual telephone customers is Voice over Asynchronous Transmission Mode using ATM Adaptation Layer 2 (VoAAL2).
Data transmitted via the VoAAL2 technology typically interfaces with a customer's equipment that uses the DS1 format. DS1 is a signal format that comprises up to 24 individual circuits called DS0s. Each DS0 uses 64 kilobits per second of bandwidth and typically contains a voice call, a fax call, or other voice-band data such as a computer modem call. A fax call requires the full 64k bandwidth. Voice calls may pass through a coding/decoding device (codec) that can perform data compression on the call. If no data compression is performed the voice call requires the full 64k bandwidth. With data compression the voice call requires less than the full 64k of bandwidth. DS0s are often grouped into trunk groups wherein all DS0s in a group have the same value of codec data compression. A trunk group supports a specific set of features and functionality. For example, a trunk group may use dual-tone multifrequency to dial digits and may provide incoming call termination only, while another trunk group may use dial pulses to dial digits and may provide outgoing call origination only.
Protocol conversion elements are typically present at a customer's premises to serve as an interface between the customer's DS1 equipment and the VoAAL2 service. Suitable conversion interfaces include the MGX 8850 manufactured by Cisco and the Passport 6480 manufactured by Nortel. A transmission of data in the VoAAL2 format will be referred to as a VoAAL2 circuit. A group of VoAAL2 circuits that is the output of a single conversion interface is known as an ATM virtual channel connection (VCC). Multiple VoAAL2 circuits are typically present in a VCC. Multiple interfaces may be present into which multiple groups of DS0s may be input and from which multiple VCCs may be output. When a group of DS0s enters a set of interfaces to be converted to the VoAAL2 format, all of the DS0s in one trunk group do not necessarily pass through the same interface. A customer may specify that some of the DS0 circuits in a trunk group should enter one interface and other circuits in the group should enter another interface. A customer can also specify the maximum number of fax calls likely to occur within a trunk group and the codec compression type for a trunk group. When multiple trunk groups are present, a different codec compression type could be used for each trunk group. If the DS0s within a trunk group are sent to different interfaces, a single interface could receive DS0s having different types of codec compression. Thus, a single VCC could carry faxes coming from different trunk groups and voice calls with different levels of data compression.
If all calls used the same bandwidth, calculating the total bandwidth on a line with multiple calls would simply be a matter of multiplying the bandwidth of each call by the number of calls. When multiple calls on a single line have disparate levels of data compression, such a straightforward calculation is not possible. The bandwidth of each call must be taken into account to determine the total bandwidth for the line. It would be desirable to have a method to determine the bandwidth needed by a VCC based on the number of fax calls and the codec types of the voice calls on the DS0s entering an interface.